One type of network that has received considerable interest over the past several years for its voice conveyance capabilities is the packet data network. In such a network, sound at an origination point may be digitized, placed into packets, and sent across the network in the packets to a destination point, which may reproduce the sound based on the data in the packets.
Unfortunately, the packets in such a network may be sent at irregular intervals, sent by different routes, and/or discarded. This leads to voice packets arriving at irregular intervals, arriving in a different order, and/or not arriving at all relative to their generation at the origination point. Thus, voice quality may suffer.
Typical systems for assessing voice quality in a packet data network require recording a test voice stream at a destination point and generating a reference voice stream from a reference voice sample. The recorded voice stream and the generated voice stream may then be compared to determine the voice quality of the network.
This approach, however, requires an additional device in the network under test so that the test voice stream may be introduced. Furthermore, introducing the test voice stream requires coordination between the origination and destination points and produces extra load in the network, which corrupts the analysis. Additionally, by only being able to measure voice quality from the origination point to the destination point, isolating problems in the network is difficult.